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<a name="farstream-plugins-FsRtpSession"></a><div class="titlepage"></div>
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<h2><span class="refentrytitle"><a name="farstream-plugins-FsRtpSession.top_of_page"></a>FsRtpSession</span></h2>
<p>FsRtpSession — A RTP session in a <a class="link" href="farstream-plugins-FsRtpConference.html#FsRtpConference" title="struct FsRtpConference"><span class="type">FsRtpConference</span></a></p>
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<a name="farstream-plugins-FsRtpSession.synopsis"></a><h2>Synopsis</h2>
<pre class="synopsis"> <a class="link" href="farstream-plugins-FsRtpSession.html#FsRtpSession" title="FsRtpSession">FsRtpSession</a>;
</pre>
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<div class="refsect1">
<a name="farstream-plugins-FsRtpSession.properties"></a><h2>Properties</h2>
<pre class="synopsis">
"<a class="link" href="farstream-plugins-FsRtpSession.html#FsRtpSession--no-rtcp-timeout" title='The "no-rtcp-timeout" property'>no-rtcp-timeout</a>" <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> : Read / Write
"<a class="link" href="farstream-plugins-FsRtpSession.html#FsRtpSession--rtp-header-extension-preferences" title='The "rtp-header-extension-preferences" property'>rtp-header-extension-preferences</a>" <span class="type">FsRtpHeaderExtensionGList</span>* : Read / Write
"<a class="link" href="farstream-plugins-FsRtpSession.html#FsRtpSession--rtp-header-extensions" title='The "rtp-header-extensions" property'>rtp-header-extensions</a>" <span class="type">FsRtpHeaderExtensionGList</span>* : Read
"<a class="link" href="farstream-plugins-FsRtpSession.html#FsRtpSession--send-bitrate" title='The "send-bitrate" property'>send-bitrate</a>" <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> : Read / Write
"<a class="link" href="farstream-plugins-FsRtpSession.html#FsRtpSession--ssrc" title='The "ssrc" property'>ssrc</a>" <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> : Read / Write
</pre>
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<div class="refsect1">
<a name="farstream-plugins-FsRtpSession.description"></a><h2>Description</h2>
<p>
This object represents one session, it is created by called
<a href="../farstream-libs-0.10/FsConference.html#fs-conference-new-session"><code class="function">fs_conference_new_session()</code></a> on a <a class="link" href="farstream-plugins-FsRtpConference.html#FsRtpConference" title="struct FsRtpConference"><span class="type">FsRtpConference</span></a>. It can be either
Audio or Video. It also represents data send with one and only one
SSRC (although if there is a SSRC collision, that SSRC may change).
</p>
<div class="refsect2">
<a name="id607651"></a><h3>Codec profiles</h3>
<p>
It is possible to define "codec profiles", that is non-autodetected
encoding and decoding pipelines for codecs. It is even possible to declare
entirely new codecs using this method.
</p>
<p>
To create a profile for a codec, add it to the codec-preferences with
special optional parameters called "farstream-send-profile" and
"farstream-recv-profile", these should contain gst-launch style descriptions
of the encoding or decoding bin.
</p>
<p>
As a special case, encoding profiles can have more than one unconnected
source pad, all of these pads should produce application/x-rtp of some kind.
The profile will be ignored if not ALL pads match currently negotiated
codecs.
</p>
<p>
Also, it is possible to declare profiles with only a decoding pipeline,
you will only be able to receive from this codec, the encoding may be a
secondary pad of some other codec.
</p>
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<p>
</p>
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<div class="refsect1">
<a name="farstream-plugins-FsRtpSession.details"></a><h2>Details</h2>
<div class="refsect2">
<a name="FsRtpSession"></a><h3>FsRtpSession</h3>
<pre class="programlisting">typedef struct _FsRtpSession FsRtpSession;</pre>
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<div class="refsect1">
<a name="farstream-plugins-FsRtpSession.property-details"></a><h2>Property Details</h2>
<div class="refsect2">
<a name="FsRtpSession--no-rtcp-timeout"></a><h3>The <code class="literal">"no-rtcp-timeout"</code> property</h3>
<pre class="programlisting"> "no-rtcp-timeout" <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> : Read / Write</pre>
<p>This is the time (in ms) after which data received without RTCP is attached the FsStream, this only works if there is only one FsStream. -1 will wait forever. 0 will not wait for RTCP and attach it immediataly to the FsStream and prohibit the creation of a second FsStream.</p>
<p>Allowed values: >= G_MAXULONG</p>
<p>Default value: 7000</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--rtp-header-extension-preferences"></a><h3>The <code class="literal">"rtp-header-extension-preferences"</code> property</h3>
<pre class="programlisting"> "rtp-header-extension-preferences" <span class="type">FsRtpHeaderExtensionGList</span>* : Read / Write</pre>
<p>GList of RTP Header extensions that are locally supported and desired by the application.</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--rtp-header-extensions"></a><h3>The <code class="literal">"rtp-header-extensions"</code> property</h3>
<pre class="programlisting"> "rtp-header-extensions" <span class="type">FsRtpHeaderExtensionGList</span>* : Read</pre>
<p>GList of RTP Header extensions that have been negotiated and will be used when sending of receiving RTP packets.</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--send-bitrate"></a><h3>The <code class="literal">"send-bitrate"</code> property</h3>
<pre class="programlisting"> "send-bitrate" <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> : Read / Write</pre>
<p>The bitrate that the session will try to send at in bits/sec.</p>
<p>Default value: 0</p>
</div>
<hr>
<div class="refsect2">
<a name="FsRtpSession--ssrc"></a><h3>The <code class="literal">"ssrc"</code> property</h3>
<pre class="programlisting"> "ssrc" <a href="http://library.gnome.org/devel/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> : Read / Write</pre>
<p>This is the current SSRC used to send data (defaults to a random value).</p>
<p>Default value: 0</p>
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